A git mirror of http://svn.asterisk.org/svn/asterisk . May lag a few hours behind. Mirrors /branches (and /trunk ). Includes tags for /tags . Does not include /team . See also it's web interface: http://svnview.digium.com/svn/asterisk . http://asterisk.org/

Russell Bryant 6e820192d6 remove misleading message vor 19 Jahren
agi a89166abbe Implement Fast AGI (agi://) over TCP socket (Astricon talk idea) vor 20 Jahren
apps 612978966a add a couple missing queue_log entries (issue #5422) vor 19 Jahren
astman c09d658263 hopefully the last try at making this happy across vor 20 Jahren
cdr 30f4cea04e attempt a restart on a connection error (bug #3628) vor 20 Jahren
channels 5835245f9f Remove ancient copy/paste error (issue #5541) vor 19 Jahren
codecs 1bde5ad687 tweak for arm4vl (bug #4545) vor 20 Jahren
configs 8a527b64dc change insecure options to support 'port' and/or 'invite' instead of forcing vor 19 Jahren
contrib 26f3eab611 This commit was manufactured by cvs2svn to create branch 'v1-0'. vor 19 Jahren
db1-ast 413c9c7009 get rid of some compile warnings (bug #2540) vor 20 Jahren
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editline 3787420fd7 Merge remaining audit patch (save dlfcn.c) vor 20 Jahren
formats d62336695b merge endian.h (bug #3867) vor 20 Jahren
images ce810cd083 Version 0.1.12 from FTP vor 23 Jahren
include 828838005a correctly fix build issues on Mac OSX Tiger by using a more generic means vor 19 Jahren
keys 182844e20d Add information for IAX on Free World Dialup vor 21 Jahren
pbx 6f2f63d22a fix callerid matching in extensions.conf vor 20 Jahren
redhat e9a32bab86 add missing line for the autosupport script (bug #3828) vor 20 Jahren
res c69e3c8393 revert SIGHUP patch to restore original behavior for 1.0 (bug #4854) vor 19 Jahren
sounds c5cf728e59 This commit was manufactured by cvs2svn to create branch 'v1-0'. vor 19 Jahren
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BUGS c9c5282183 Update Changelog/BUGS vor 20 Jahren
CREDITS 4f26f4ea63 add Rich Murphey to the CREDITS vor 20 Jahren
ChangeLog 6e820192d6 remove misleading message vor 19 Jahren
HARDWARE 39c8a208c0 Plane commits (a.k.a. the Delta deltas): 1) Make muted reconnect 2) Add "X" option to meetme and add ${MEETME_EXIT_CONTEXT}, 3) Allow SIP call parking with supervised transfer, 4) Only create parking entries when calls actually get parked, 5) Add "sunshine" song, 6) Update hardware documentation, 7) Don't load empty strings from history file vor 20 Jahren
LICENSE 99a14e4a20 Version 0.1.1 from FTP vor 25 Jahren
Makefile 34ca4907e5 re-add explicit poll support for Darwin vor 19 Jahren
README a853b8de56 add notes about file descriptors (bug #4134) vor 20 Jahren
README.fpm ebf2527ed9 Add little note about hold music vor 20 Jahren
SECURITY 4369bc6de7 Update security document, work on threading with pbx.c and small SIP fixes vor 21 Jahren
acl.c e55c70b25e ensure that calls to gethostbyname are null terminated, vor 20 Jahren
aescrypt.c 5f99d0a46c Add AES support vor 21 Jahren
aeskey.c 5f99d0a46c Add AES support vor 21 Jahren
aesopt.h d62336695b merge endian.h (bug #3867) vor 20 Jahren
aestab.c 5f99d0a46c Add AES support vor 21 Jahren
alaw.c 999196dfe0 Version 0.1.10 from FTP vor 23 Jahren
app.c 3a75ee0f7e Actually write audio to file in get_voice (issue #5547) vor 19 Jahren
ast_expr.y c1464d4b9a Fix setvar issue (bug #3010) vor 20 Jahren
astconf.h f34ddecce6 Version 0.3.0 from FTP vor 22 Jahren
asterisk.8.gz 28b70914e3 Add Asterisk manpage vor 20 Jahren
asterisk.c c696c5bf84 only initialize random number generators in one place (bug #4017) vor 20 Jahren
asterisk.h 6138785b1f Change recent patch to not use a hard coded path inside of a .c file vor 20 Jahren
asterisk.sgml 28b70914e3 Add Asterisk manpage vor 20 Jahren
astmm.c 4f9cfda67d Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). vor 21 Jahren
autoservice.c 0e9e565ef1 Merge BSD stack size work (bug #2067) vor 20 Jahren
callerid.c e4169f823f Ignore message type in CID Delivery (bug #2552) vor 20 Jahren
cdr.c eb10e579f1 Fix CDR for supervised transfer in chan_zap and chan_sip (bug #1595) vor 20 Jahren
channel.c 6b315d4072 copy the monitor over when masquerading (bug #3809) vor 19 Jahren
chanvars.c 5cb14f5a82 Include fixes for portability vor 22 Jahren
cli.c bdab063a07 fix incorrect CLI tab completion (issue #5041) vor 19 Jahren
coef_in.h ff9afb76c9 Merge UK + DTMF Caller*ID stuff and fix app_test description vor 20 Jahren
coef_out.h 5f36e14213 Version 0.1.7 from FTP vor 24 Jahren
config.c e294aa18d5 fix ast config path (bug #4184) vor 20 Jahren
db.c d7757824f6 More strcpy / snprintf as part of rgagnon's audit (bug #2004) vor 20 Jahren
dlfcn.c 500266532a fix misspelling of separate (bug #3607) vor 20 Jahren
dns.c d62336695b merge endian.h (bug #3867) vor 20 Jahren
dsp.c d7b76a8e2e update unused code ... (bug #3342) vor 20 Jahren
ecdisa.h 37c7d62218 Version 0.1.10 from FTP vor 23 Jahren
enum.c a6e5351743 ensure buffers are large enough for ENUMLookup (issue #4943) vor 19 Jahren
file.c fc977dee1f fix return values on systems where an unsigned char is the default (bug #4455) vor 20 Jahren
frame.c 73f95deadb fix queue URL passing (bug #3543) vor 20 Jahren
fskmodem.c ff9afb76c9 Merge UK + DTMF Caller*ID stuff and fix app_test description vor 20 Jahren
image.c 4f9cfda67d Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). vor 21 Jahren
indications.c 500266532a fix misspelling of separate (bug #3607) vor 20 Jahren
io.c bbd0abbff6 Make it build and run on MacOS X vor 21 Jahren
loader.c 828838005a correctly fix build issues on Mac OSX Tiger by using a more generic means vor 19 Jahren
logger.c 2032743856 make sure an automatic log rotation doesn't result in nasty recursion (bug #4646) vor 19 Jahren
make_build_h 2fc1e33737 Version 0.1.8 from FTP vor 24 Jahren
manager.c d5f7e52368 Fix poll error condition causing memory corruption (bug #4915) vor 19 Jahren
md5.c d62336695b merge endian.h (bug #3867) vor 20 Jahren
mkdep dad6895322 Make mkdep throw away stderr since people think the error messages printed are serious when they are not vor 21 Jahren
muted.c 39c8a208c0 Plane commits (a.k.a. the Delta deltas): 1) Make muted reconnect 2) Add "X" option to meetme and add ${MEETME_EXIT_CONTEXT}, 3) Allow SIP call parking with supervised transfer, 4) Only create parking entries when calls actually get parked, 5) Add "sunshine" song, 6) Update hardware documentation, 7) Don't load empty strings from history file vor 20 Jahren
muted.conf.sample 04c4adfdfa clean up config file sample vor 21 Jahren
pbx.c 1f7dcd6528 spell them words cowrecktly (issue #4964) vor 19 Jahren
poll.c 8dea70a34c Make it build and run on MacOS X vor 21 Jahren
privacy.c 4f9cfda67d Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). vor 21 Jahren
rtp.c 42910b53d1 prevent possible seg fault (issue #5502) vor 19 Jahren
sample.call 435a6fe182 Add example of using Account in sample.call file vor 21 Jahren
say.c 3652080f86 fix 'say phonetic' (issue #5268) vor 19 Jahren
sched.c 82ba51b02f create useful output for time left to expire (bug #4022) vor 20 Jahren
sounds.txt 82f451000c add hello world prompt to 1.0 for Jared vor 19 Jahren
srv.c 33dbddcc3f REduce chattyness vor 20 Jahren
tdd.c df52bd2463 Backport recent memory fixes to 1.0 vor 20 Jahren
term.c 0ba369c714 Support colors in eterm vor 20 Jahren
translate.c 55aa5114a2 Rename newp to newpvt (bug #2190), change hold music. vor 20 Jahren
ulaw.c ba35830b60 Version 0.1.10 from FTP vor 23 Jahren
utils.c ca4585354d fix crash on amd64 (issue #5210) vor 19 Jahren

README

The Asterisk Open Source PBX
by Mark Spencer
Copyright (C) 2001-2004 Digium
================================================================
* SECURITY
It is imperative that you read and fully understand the contents of
the SECURITY file before you attempt to configure an Asterisk server.

* WHAT IS ASTERISK
Asterisk is an Open Source PBX and telephony toolkit. It is, in a
sense, middleware between Internet and telephony channels on the bottom,
and Internet and telephony applications at the top. For more information
on the project itself, please visit the Asterisk home page at:

http://www.asterisk.org

In addition you'll find lot's of information compiled by the Asterisk
community on this Wiki:

http://www.voip-info.org/wiki-Asterisk

* LICENSING
Asterisk is distributed under GNU General Public License. The GPL also
must apply to all loadable modules as well, except as defined below.

Digium, Inc. (formerly Linux Support Services) retains copyright to all
of the core Asterisk system, and therefore can grant, at its sole discretion,
the ability for companies, individuals, or organizations to create proprietary
or Open Source (but non-GPL'd) modules which may be dynamically linked at
runtime with the portions of Asterisk which fall under our copyright
umbrella, or are distributed under more flexible licenses than GPL.


If you wish to use our code in other GPL programs, don't worry -- there
is no requirement that you provide the same exemption in your GPL'd
products (although if you've written a module for Asterisk we would
strongly encourage you to make the same exemption that we do).

Specific permission is also granted to OpenSSL and OpenH323 to link to
Asterisk.

If you have any questions, whatsoever, regarding our licensing policy,
please contact us.

Modules that are GPL-licensed and not available under Digium's
licensing scheme are added to the Asterisk-addons CVS module.

* REQUIRED COMPONENTS

== Linux ==
Currently, the Asterisk Open Source PBX is only known to run on the
Linux OS, although it may be portable to other UNIX-like operating systems
(like FreeBSD) as well.


* GETTING STARTED

First, be sure you've got supported hardware (but note that you don't need ANY hardware, not even a soundcard) to install and run Asterisk. Supported are:

* All Wildcard (tm) products from Digium (www.digium.com)
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
* Full Duplex Sound Card supported by Linux
* Adtran Atlas 800 Plus
* ISDN4Linux compatible ISDN card
* Tormenta Dual T1 card (www.bsdtelephony.com.mx)

Hint: CAPI compatible ISDN cards can be run using the add-on channel chan_capi.

So let's proceed:

1) Run "make"
2) Run "make install"

If this is your first time working with Asterisk, you may wish to install
the sample PBX, with demonstration extensions, etc. If so, run:

"make samples"

Doing so will overwrite any existing config files you have. If you are lacking a soundcard you won't be able to use the DIAL command on the console, though.

Finally, you can launch Asterisk with:

./asterisk -vvvc

You'll see a bunch of verbose messages fly by your screen as Asterisk
initializes (that's the "very very verbose" mode). When it's ready, if
you specified the "c" then you'll get a command line console, that looks
like this:

*CLI>

You can type "help" at any time to get help with the system. For help
with a specific command, type "help ". To start the PBX using
your sound card, you can type "dial" to dial the PBX. Then you can use
"answer", "hangup", and "dial" to simulate the actions of a telephone.
Remember that if you don't have a full duplex sound card (And asterisk
will tell you somewhere in its verbose messages if you do/don't) than it
won't work right (not yet).

Feel free to look over the configuration files in /etc/asterisk, where
you'll find a lot of information about what you can do with Asterisk.

* ABOUT CONFIGURATION FILES

All Asterisk configuration files share a common format. Comments are
delimited by ';' (since '#' of course, being a DTMF digit, may occur in
many places). A configuration file is divided into sections whose names
appear in []'s. Each section typically contains two types of statements,
those of the form 'variable = value', and those of the form 'object =>
parameters'. Internally the use of '=' and '=>' is exactly the same, so
they're used only to help make the configuration file easier to
understand, and do not affect how it is actually parsed.

Entries of the form 'variable=value' set the value of some parameter in
asterisk. For example, in tormenta.conf, one might specify:

switchtype=national

In order to indicate to Asterisk that the switch they are connecting to is
of the type "national". In general, the parameter will apply to
instantiations which occur below its specification. For example, if the
configuration file read:

switchtype = national
channel => 1-4
channel => 10-12
switchtype = dms100
channel => 25-47

Then, the "national" switchtype would be applied to channels one through
four and channels 10 through 12, whereas the "dms100" switchtype would
apply to channels 25 through 47.

The "object => parameters" instantiates an object with the given
parameters. For example, the line "channel => 25-47" creates objects for
the channels 25 through 47 of the tormenta card, obtaining the settings
from the variables specified above.

* SPECIAL NOTE ON TIME

Those using SIP phones should be aware the Asterisk is sensitive to
large jumps in time. Manually changing the system time using date(1)
(or other similar commands) may cause SIP registrations and other
internal processes to fail. If your system cannot keep accurate time
by itself use NTP (http://www.ntp.org/) to keep the system clock
synchronized to "real time". NTP is designed to keep the system clock
synchronized by speeding up or slowing down the system clock until it
is synchronized to "real time" rather than by jumping the time and
causing discontinuities. Most Linux distributions include precompiled
versions of NTP. Beware of some time synchronization methods that get
the correct real time periodically and then manually set the system
clock.

Apparent time changes due to daylight savings time are just that,
apparent. The use of daylight savings time in a Linux system is
purely a user interface issue and does not affect the operation of the
Linux kernel or Asterisk. The system clock on Linux kernels operates
on UTC. UTC does not use daylight savings time.

Also note that this issue is separate from the clocking of TDM
channels, and is known to at least affect SIP registrations.

* FILE DESCRIPTORS

Depending on the size of your system and your configuration,
Asterisk can consume a large number of file descriptors. In UNIX,
file descriptors are used for more than just files on disk. File
descriptors are also used for handling network communication
(e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and
digital trunk hardware). Asterisk accesses many on-disk files for
everything from configuration information to voicemail storage.

Most systems limit the number of file descriptors that Asterisk can
have open at one time. This can limit the number of simultaneous
calls that your system can handle. For example, if the limit is set
at 1024 (a common default value) Asterisk can handle approxiately 150
SIP calls simultaneously. To change the number of file descriptors
follow the instructions for your system below:

== PAM-based Linux System ==

If your system uses PAM (Pluggable Authentication Modules) edit
/etc/security/limits.conf. Add these lines to the bottom of the file:

root soft nofile 4096
root hard nofile 8196
asterisk soft nofile 4096
asterisk hard nofile 8196

(adjust the numbers to taste). You may need to reboot the system for
these changes to take effect.

== Generic UNIX System ==

If there are no instructions specifically adapted to your system
above you can try adding the command "ulimit -n 8192" to the script
that starts Asterisk.


* MORE INFORMATION

See the doc directory for more documentation.

Finally, you may wish to visit the web site and join the mailing list if
you're interested in getting more information.

http://www.asterisk.org/index.php?menu=support

Welcome to the growing worldwide community of Asterisk users!

Mark Spencer